Re: [asterisk-users] Message waiting question...
Anyhow, Asterisk1 and Asterisk2 are connected using IAX2. What I would like is to have the SPA3000 Message Waiting indicator based on the voicemail message hosted on the Asterisk2 server. There is this old patch that does remote MWI over IAX (among other things). I used it on earlier versions and it worked quite nicely. This was before 1.2 so it may no longer work at all. At the very least it will likely required some updating. Doable, just depends how much time you want to put into it :). See: http://bugs.digium.com/view.php?id=4371 --Luki ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] OFF-TRACK: Is VOIP -PSTN integration legal in China
Hi Dan, Thanks for the reply. I was just searching the MII china site to confirm this. But could not find any information on the same. They just kept talking about issuing VOIP licences to service providers But no focus on whether an enterprise can do it for itself or not. But just one point, Do the PSTN service provider made some checks on what have you installed? I too agre with you about India. Sad..isn't it? Regards dashy --- Dan Austin [EMAIL PROTECTED] wrote: A few years back China relaxed the rules quite a bit. I'm not sure if they require that the PSTN interface cards be certified, but I have connected a number of offices in China to the PSTN using Cisco VoIP gear. Since I had no desire to visit a Chinese penal institution, nor wished that on my Chinese co-workers, I made sure to get the telcos to explain the status. The telcos were not selling gear, only services, so there was no reason not to believe them when they said it was now permissible. Too bad I cannot say the same for India... Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dashy dude Sent: Wednesday, July 26, 2006 10:44 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] OFF-TRACK: Is VOIP -PSTN integration legal in China Dear All, Can anyone tell me if I can legally integrate Asterisk with PSTN network in China. Thanks in advance dashy __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reload of wct4xxp without restarting of Asterisk?
Hello, is it possible to restart the wct4xxp kernel module and start again without stopping Asterisk? i tried to unload chan_zap.so but rmmod says the module is in use. Is it possible? if it is, how its possible? Thanks Nico ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reload of wct4xxp without restarting of Asterisk?
On Thu, 2006-07-27 at 09:04 +0200, [EMAIL PROTECTED] wrote: is it possible to restart the wct4xxp kernel module and start again without stopping Asterisk? Yes, you should be able to unload chan_zap.so from Asterisk without stopping the rest. i tried to unload chan_zap.so but rmmod says the module is in use. To unload the module, there can not be any active calls. -- Russell Bryant Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing timer
- Message d'origine
De: Mojo with Horan Company, LLC [EMAIL PROTECTED]
A: Zenone [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Objet: Re: [asterisk-users] Ringing timer
Date: 26/07/06 22:39
Yes, on a Zap FXO channel, when you can hear ringing, the timeout is
counting down, even if the remote party hasn't answered yet.
Thanks!
But I don't understand why, when I wrote this:
exten = _0X,2,Dial(${TRUNK}/${NUMPH},5,H|g)
the called phone rings more than 5 seconds and finally goes on voicemail?
Message sent using UebiMiau 2.7.8
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Re: [asterisk-users] ACD Queues Agents logout
I didn't want to send the Agent thru the whoule AgentCallbackLogin rutine just to _log off_. This does not make really sense to me. thank for your answer anyway. Kai Here is what I do... Exten=777,1,AgentCallbackLogin() ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi Asterisk Server to relay call request
Dear Gurus, I'm newbe in Asterisk and I want to evaluate the system. I have several location branch office and I want to use VOIP between them. Is there any documentation about Asterisk that cover several location and the dial plan? Is it possible to have one central Asterisk to control all the remote asterisk? Regards, Fadjar T ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reload of wct4xxp without restarting of Asterisk?
As i writed, i do an unload of chan_zap.so but i can't unload the module wct4xxp, is this possible? Thanks Nico On Thu, 27 Jul 2006, Russell Bryant wrote: On Thu, 2006-07-27 at 09:04 +0200, [EMAIL PROTECTED] wrote: is it possible to restart the wct4xxp kernel module and start again without stopping Asterisk? Yes, you should be able to unload chan_zap.so from Asterisk without stopping the rest. i tried to unload chan_zap.so but rmmod says the module is in use. To unload the module, there can not be any active calls. -- Russell Bryant Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR dest question
Hi, we are having some trouble with CDR records. Example: Case 1: Customer 12345 calls extension 10. Extension 20 takes the call using Pickup (e.g. *810). I now have two CDRs: 1: 12345 - 10 2: 20 - *810 I could live with the second CDR but the first gives the impression as if 12345 was talking to 10 while in real life he/she was talking to 20. How can I fix this? Case 2: We are using Dial commands with several channels e.g. Dial(Zap/10Zap/20). I cannot see what channel (and therefore the associated default extension with that channel) picked up the call. Is there a way to fix this? Kind regards, JP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] french promt
Please any one knows from where I can download asterisk French sounds /var/lib/asterisk/sounds. Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Error when calling
Dear Anthony, The dial plan is currently very simple it should pick up any call and send it to a sip phone registered, you can see the context below named zap-in is what I am using, it is only that and nothing more, is there something extra I have to add to dial plan or to that context ? Thx MAG Anthony Rodgers wrote: This looks like a dialplan problem - do you have a match for 0109687348 in the zap-in context in your dialplan? A. On 26-Jul-06, at 2:40 PM, Mohamed A. Gombolaty wrote: > Dear All, > I have a strange problem in recieving calls on the pri the zaptel > is green and everything seems very well, but when a call comes I > can see the call along with the caller ID but then I get this > strange message which make the call hungup: > > > error msg: 'zap-in' from '0109687348' does not exist. Rejecting > call on channel 0/18, span 1. > > the PRI is an E1 and I have the following configuration for > extensions.conf > > [zap-in] > exten => s,1,Answer > exten => s,2,Dial(sip/100) > exten => s,3,Hungup > > as for the zapata.conf it is as follow: > > [channels] > language=en > switchtype=euroisdn > signalling=pri_cpe > context=zap-in > group=0 > channel=>1-15,17-31 > > I don't know what the problem is or where to look, I will > appreciate it if someone can help me out? > > Thx > MAG > > -- Thx MAG > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip phone settings set when user registers
Hi all ! I am planing to set up around 20 SIP Phones which will be purchased in one bunch, I am more or less free of choice. I wonder if anyone knows sip phones which allow configuration upon login. The following scenario: User logs into any phone and the settings of the phone are always the same. Meaning individual key assignement is always the same. Is this possible with asterisk in combination which any phone or do I require special phones. Thanks for any advices Nik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [asterisk-users] Sip phone settings set when user registers
Hello Just use Snom or grandstream phones. They can be provisioned very easily via HTTP. You just setup a config URL on the phones, and they get their configurations from there. If you want to get more advanced, they can send along their MAC address, and thereby enabling you to custom config them directly from a central application, based on the phones MAC address. The snom phones can even be instructed to download a configuration from a URL via DHCP. Jon -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Nik Engel Sendt: 27. juli 2006 10:39 Til: asterisk-users@lists.digium.com Emne: [asterisk-users] Sip phone settings set when user registers Hi all ! I am planing to set up around 20 SIP Phones which will be purchased in one bunch, I am more or less free of choice. I wonder if anyone knows sip phones which allow configuration upon login. The following scenario: User logs into any phone and the settings of the phone are always the same. Meaning individual key assignement is always the same. Is this possible with asterisk in combination which any phone or do I require special phones. Thanks for any advices Nik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.4/401 - Release Date: 26-07-2006 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.4/401 - Release Date: 26-07-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Recording/Monitor after xfer
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Telles Rodrigo P. Telles wrote: Hi, I'd like to know if some one knows how to make Asterisk record a call after xfer (not bxfer). I tried some ways but it doesn't work at all. I assume that you are using Asterisk 1.0.X. - I had the same problem until upgrading to 1.2 - -- Ron Wellsted [EMAIL PROTECTED] http://www.wellsted.org.uk N 52.567623, W 2.137621 Linux Counter No. 202120 FWD:519961 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2.2 (GNU/Linux) iQEVAwUBRMiBiktP/KMNOfRbAQLpPQgArUdVUNGzejClFa37a7ppcx0e+VWJNTT7 QjqdfeZFzexaKqlyicCpyicFG0oVUDKg+M6oOZpJYBEmka40Pvt/+tkaQ3pqvLT4 eLZ74svakQ5c9C86kvnwJWBkkC4wnMafaWKpAJNgtvihbOcT9J4jmYm3jkqPM5ha lHhMovDh6O3ba+hox/iAkqCddtN0U3EcUyPEM6ZHMCX+xFeDjU0f1cpmbZZ9I4nN NIBeseEzOX1VmU/CHqnQsCdQGvV3gpab10uEL0p0DQzwSoMJI81vLeC8FF4odc9d TsmW92+c42uLadSk7CdWiYp7jWdLGzgXX2IU5MpvBm3PIllHBQxILw== =IeJA -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip phone settings set when user registers
Hi Nik, I like the Grandstream Budge Tone 102 VoIP Phones which you can find here: http://www.voipsupply.com/product_info.php?products_id=40 and here: http://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-31609737728.htm Also, the GXP-2000 is a very popular model too, although once you consider the capabilities of Asterisk the only real advantage this unit has over the others (even in an office environment), is the Power over Ethernet (PoE) feature: Nik Engel wrote: Hi all ! I am planing to set up around 20 SIP Phones which will be purchased in one bunch, I am more or less free of choice. I wonder if anyone knows sip phones which allow configuration upon login. The following scenario: User logs into any phone and the settings of the phone are always the same. Meaning individual key assignement is always the same. Is this possible with asterisk in combination which any phone or do I require special phones. Thanks for any advices Nik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] french promt
Hello Khaled, Follow this link: http://svn.digium.com/view/asterisk/sounds/fr/trunk/?rev=34575 Best regards, Olivier S. Khaled Chehab a écrit : Please any one knows from where I can download asterisk French sounds /var/lib/asterisk/sounds. //Regards// * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Olivier Saulnier STEGANUX 1er étage DIAMECANS BEL AIR 03410 St-Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mobile SIP Client
Dear All, I'm looking for a mobile SIP client to use with Asterisk. Has anyone got experience in this area and can you advise me of a product? Many Thanks Shad Mortazavi -- Nexus Group Technical Manager n|m Nexus Management plc SIP: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [oh323]FastStart/H245Tunnelling/H245inSetup
Hi all, i have following setup []--[asterisk]--[oh323]--[HiPath]--[8000] is my voicemail access exten = ,1,Answer() exten = ,2,VoiceMailMain() 8000 is an Optiset phone registered on the HiPath. When 8000 calls i have no voice (depends on the setting of FastStart). When FastStart=yes in oh323 the caller can't hear the voivemail message (otherwise (when FastStart=no) every thing works fine. Can anyone explain the impact of FastStart? What is the H245inSetup parameter? thx in advance... __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Malformed/Missing URL Problem with Cisco Callmanager 4.1
Hi I want to use Asterisk as a Voicemail Box for my Callmanager Users The Link between Cisco Callmanager and Asterisk has to be SIP (according to http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration) The Voicemail Part on Asterisk is running perfect via a IAX Softphone but not via the SIP Channel (SIP Trunk in Cisco words) The Callmanager Box and the Asterisk Box are on the same Subnet/VLAN - there is no Firewall or something else between them I am always getting this Error on the Asterisk CLI : -- SIP read from 10.200.16.52:5060: SIP/2.0 400 Bad Request - 'Malformed/Missing URL' Via: SIP/2.0/UDP 10.200.16.72:5060;branch=z9hG4bK0b0171ec;rport From: asterisk sip:[EMAIL PROTECTED];tag=as027c0ecb To: sip:callmanagertest.firm.country Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Content-Length: 0 --- (7 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' Asterisk Versions I tried : 1.2.7 - 1.2.10 Callmanager Versions I tried : 4.1 - 4.2.1sr1a Changing the Version of Asterisk or Callmanager doesn't help. So I think the Problem is in my Asterisk SIP Trunk Configuration. At the moment the configuration looks like : [general] context=default allowguest=no realm=tds.de bindport=5060 bindaddr=10.200.16.72 srvlookup=no autodomain=yes domain=firm.country domain=10.200.16.52 vmexten=voicemail videosupport=no disallow=all allow=ulaw allow=alaw relaxdtmf=yes rtptimeout=60 rtpholdtimeout=300 useragent=Asterisk dtmfmode=rfc2833 sipdebug=yes notifyringing=yes [default] include = callmanager2-1 include = callmanager2-2 [callmanager2-1] type=friend context=default host=callmanagertest.firm.country dtmfmode=rfc2833 port=5060 insecure=port,invite disallow=all allow=ulaw allow=alaw nat=no canreinvite=yes username=phone fromuser=phone qualify=yes [callmanager2-2] type=friend context=default host=callmanagertest.firm.country dtmfmode=rfc2833 port=5060 insecure=port,invite disallow=all allow=ulaw allow=alaw nat=no canreinvite=yes username=phone fromuser=phone qualify=yes Has anyone any Idea ? :) or perhaps some Sample Configuration Files of such a scenario ? Many thanks David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip phone settings set when user registers
Hi ! Also, the GXP-2000 is a very popular model too, although once you consider the capabilities of Asterisk the only real advantage this unit has over the others (even in an office environment), is the Power over Ethernet (PoE) feature: which is supported be Snoom as well. Anyway I would be more interested in a method to configure the key assignements upon login with asterisk ?? any ideas how to do that ? Nik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Nokia E61/E70 not always answering voip calls
Has anyone else had problems with the Nokia E61 and E70 phones not always answering voip calls? We have them connected via a local access point (so no router/NAT) and sometimes the phones dont ring when called. They are registered ok and if you use the phone to make a voip call it works fine. The last time someone called me I answered the call on my desk phone and a few seconds later the mobile rang and continued to ring even after I hung up the call on my desk phone! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] alcatel ip touch 4068 ... sip?
Hi, Quickie ... is the alcatel ip touch 4068 (or any other in that series) sip enabled? If not, does alcatel have a sip-enabled phone? Cesc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue announcement issues
On 07/27/06 03:28 Phil Jordan said the following: Jul 26 20:05:22 DEBUG[16371] channel.c: Scheduling timer at 160 sample intervals Jul 26 20:05:22 DEBUG[16371] channel.c: Avoiding initial deadlock for 'IAX2/phil-5' Jul 26 20:05:22 VERBOSE[16371] logger.c: -- Called IAX2/phil Jul 26 20:05:22 DEBUG[16371] channel.c: Generator got voice, switching to phase locked mode Jul 26 20:05:22 DEBUG[16371] channel.c: Scheduling timer at 0 sample intervals Jul 26 20:05:22 VERBOSE[16371] logger.c: -- Call accepted by 82.11.45.110 (format gsm) Jul 26 20:05:22 VERBOSE[16371] logger.c: -- Format for call is (gsm) Jul 26 20:05:22 VERBOSE[16371] logger.c: -- IAX2/phil-5 is ringing Jul 26 20:05:56 DEBUG[16371] chan_sip.c: Stopping retransmission on it does seem that IAX2/phil is still logged in as an agent of the queue, thus the caller is delivered to that agent and no hold times, position or periodic announcements are made. what does 'show queue hasbean' and 'show agents' say ? this may be the case because you have persistentagents=yes in queues.conf. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nokia E61/E70 not always answering voip calls
On 7/27/06, Gareth Blades [EMAIL PROTECTED] wrote: Has anyone else had problems with the Nokia E61 and E70 phones not always answering voip calls? We have them connected via a local access point (so no router/NAT) and sometimes the phones dont ring when called. They are registered ok and if you use the phone to make a voip call it works fine. The last time someone called me I answered the call on my desk phone and a few seconds later the mobile rang and continued to ring even after I hung up the call on my desk phone! That is wireless VoIP for you :( I would suggest enabling qualify=yes in your sip.conf in order to get an idea of the quality of the wireless link from Asterisk's perspective. I have found it quite revealing in the past, and generally not in a good way... Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip phone settings set when user registers
On 7/27/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello Just use Snom or grandstream phones. They can be provisioned very easily via HTTP. You just setup a config URL on the phones, and they get their configurations from there. If you want to get more advanced, they can send along their MAC address, and thereby enabling you to custom config them directly from a central application, based on the phones MAC address. The snom phones can even be instructed to download a configuration from a URL via DHCP. This is true, but making a change to the configuration (pushing a change) based on a user action is much harder, or even impossible. We have compromised by having the phones configured once at boot, and having Asterisk change the behaviour under the hood when the user requests it. Not tidy, but the end result works. Cheers, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Manager interface
Title: Manager interface This has probably been discussed before but I need to do a screen pop and I'm looking for ways to do it. I am assuming I need to use the manager interface, which is ok cos I'm using that for calling out but I'm not quite what to pick up on. Regards Lee ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Woes
Hi Dave The problem is with the way in which Asterisk handles 'overlap' dialling with SIP. i.e. not very well at all. If you remove the early dial feature from the phone I think you will find it will solve the problem. The issue is that Asterisk does not apply the digit timeout on SIP early dial. To behave like chan_zap does with overlap dialling the timeout MUST be in Asterisk. I have had a long email exchange with Olle, master of chan_sip but he told me that SIP was never designed to operate in this manner for overlap dialling. If so, I think this is actually a major flaw in SIP...Unfortunately multi-length dialplans are a fact of life in many countries, which makes SIP non-optimal. Rgds Tim Robinson Basingstoke, UK Dave Hope wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello all, I've been trying to play with asterisk (after a two month break) and am having some problems getting my SIP connection to a third party provider to work. In the asterisk console I notice: - - debian*CLI set verbose 999 Verbosity was 0 and is now 999 Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:2355 sip_alloc: Allocating new SIP call for [EMAIL PROTECTED] Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:5441 check_user_full: Setting NAT on RTP to 4 Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:840 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 1: Found Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:5441 check_user_full: Setting NAT on RTP to 4 Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:7329 handle_request: Check for res for 200 Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:1620 update_user_counter: Call from user '200' is 1 out of 0 Jul 23 16:40:51 DEBUG[4043]: chan_sip.c:840 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 2: Found - - I believe that's some sort of SIP routing issue related to ReInvite's ? - - Is there a workaround for this? In the attempt that someone may be able to shed some light on the matter, I've uploaded my current configuration to: http://files.davehope.co.uk/asterisk-problem/ I've also uploaded the output of 'sip debug'. The interesting bit in that (to me at least) is the message: - - Looking for 10 in Outgoing Reliably Transmitting (NAT): SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP - - Is it so simple that I've missed something out in my outgoing bit on my dialplan ? Anyway, the complete log can be found here: http://files.davehope.co.uk/asterisk-problem/debug.log Ohh. And: - - [EMAIL PROTECTED]:/etc/asterisk# asterisk -V Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k - - If anyone would be so kind as to shed some insight into the matter it'd be greatly appreciated!, Kind Regards, Dave -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.4 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEw4ywjdL3ZT1KDlERAvgqAJ9ptCZlpKeFDkdKNaOHBKDLHi3HrgCglG3I 5K48wq9FfL4VlBkADOtLvXU= =57su -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Manager interface
Title: Manager interface If you want to do a screen popup when an agent receives a call, then you should consider looking at these events: AgentCalled AgentConnect AgentComplete p.s: I'm not sure, but you might need to set eventmemberstatus=yes in your queue.conf to receive these events From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer Sent: Thursday, July 27, 2006 12:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Manager interface This has probably been discussed before but I need to do a screen pop and I'm looking for ways to do it. I am assuming I need to use the manager interface, which is ok cos I'm using that for calling out but I'm not quite what to pick up on. Regards Lee ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dropping calls in the middle of conversation
Hi all, I'm having major trouble with a simple asterisk installation dropping calls in the middle of the conversation. I recentlyupgraded from asterisk-1.2.3 and zaptel-1.2.2 to asterisk 1.2.10 and zaptel-1.2.7, but to no avail. The machine is equiped with a TDM40B and a TDM22B and has an MSI motherboard with an intel 915G chipset and a SATA hard disk 80 GB. Its running Fedora core 3 Linux. The situation appears with no obvious reason, the CLI shows nothing more than the zaptel channel hanging up. How should i go debugging this mess? Giannis Margaritis ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Determining what gets written to the dst field for a CDR
I use CDR(userfield) to store dialed numbers
[call9]
exten = s,2,Set(CDR(userfield)=${number})
exten = s,3,Dial(Zap/2/${number}|40)
[outgoing]
exten = _9.,1,Set(number=${EXTEN:1})
exten = _9.,2,Goto(call9|s|1))
in cdr i have
| dst |userfield
+-+-
| s | ${number}
I have Asterisk set up to write call detail records to MySQL. The
number written to the dst field is the number dialled by the user
including any prefix (e.g. 12125554433 where 1 gives an outside line).
However this is not the number dialled by Asterisk (e.g. in this case
Asterisk would drop the 1 and dial 2125554433). Is it possible to
write the CDR record with the number dialled by Asterisk rather than
that dialled by the user?
Any advice appreciated.
Regards
Cameron
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Re: [asterisk-users] sip realtime
Ciao Benchev, Also register= can be done only from a .conf file. Well, I'm experimenting right now with this, and I can tell you that register = works even with static realtime. HTH, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Rxfax and squashed TIFF
Hi All Just wondering if anyone knows of a solution to the squashed tiff problem with spandsp (or rather Windows Image Viewer) other than converting to a PDF. I find the PDF image quality is not nearly as good as the original TIFF. Apparently the Windows Image Viewer doesn't understand the metric units in the TIFF header. Convert them to imperial units? Kind Regards Garth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] HP DL380 and the TE4xxP cards
Sangoma 104D on a DL320, 3ghz, 1gig ram, NFAS, 50% CPU utilization @ 95 calls. Only running asterisk and passing calls off via ulaw SIP. Thanks, Steve Totaro -Original Message- From: Patrick [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 26, 2006 9:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HP DL380 and the TE4xxP cards On Thu, 2006-07-27 at 10:06 +1000, Edwin Groothuis wrote: Hello, Does anybody have experience with the Quad T1/E1 PRI cards in an HP DL380? Just a yes it works fine or a never again is enough :-) It works fine with a TE210P card. I did turn off the hyperthreading. Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] playing a sound into a meetme conf
Hi All,I have a problem and I'm not sure if a solution is possible without using the asterisk testing code.I am developing a volunteer translation service that users can dial into. I have a list of volunteer translators cell phone numbers stored in a mysql database along with times that they have volunteered to act as translators. That I pull from using some perl AGI scripts. A user calls, I ask which language they need help/translation with, then I put the users into a meetme conference while I call translators and play them a message asking if they're available at this time. They can refuse or accept the call. Once I get a translator that has accepted the call I connect the translator as an administrator to the meetme conference that is holding the user that is listening to music on hold.That is all working quite well with the Dialplan and AGI scripts I have set up. Problems happen when the translator drops the call midway through the conversation. i.e. Losing cell phone service.When that happens I need a way to play a message to the user to let them know that the translator has been lost and we're looking for a new one. I then need to put back the music on hold, then run deadagi scripts to find a new translator to connect to the meetme conference to help out the user.What is currently happening is that the user is left in the conference alone forever listening to MOH. I think there are two ways to do this, but I can't find out how to do either from any documentation I've found.1. Break the user out of the meetme conf and back into the dialplan. - If I kick them from the conference they are immediately hung up on and I don't know how to stop this from happening. - There is function that is available in Asterisk 1.4 called ManagerRedirect that seems like it could do this for me, but i'd rather not try to integrate this into 1.2.10 because I fear breaking too many other things and running 1.4 (testing) just isn't an option at this time.(details here: http://bugs.digium.com/view.php?id=6508)2. Play a message into the conference - Can I join a new pseudo channel that I've created to a meetme conf that plays a message? Does anyone know how to do this? - Can I override the MOH and stream a recorded message into the conference with only the single user in the meetme conf?Any help/ideas are appreciated.Cheers, - Simon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with call receiving (Asterisk+PSTN+Digium TDM04B)
Hi Friends, I am Chandra from India. Thank you for your cooperation and for clear my doubts. Now, I have installed Digium TDM04B card in my Asterisk server and configured. I have one landline number from PSTN. Now, I have connected that PSTN cable to my TDM04B first port. When I am making calls from outside to my PSTN number, sometimes Asterisk receving that call and sometimes, its not receiving. Why? Here I am giving my configuration of my files. ZAPTEL.CONF contents: loadzone = us defaultzone=us fxsks=1,2,3,4 ZAPATA.CONF contents: [channels] context=tutorial signalling=fxs_ks busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes cancallforward=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callerid=asreceived language=en usecallerid=yes echocancel=yes transfer=yes immediate=no group=1 channel = 1 SIP.CONF contents: [300] type=friend username=300 secret=server callerid="Server" host=dynamic context=tutorial [general] port=5060 bindaddr=0.0.0.0 context=default disallow=all allow=gsm allow=ulaw allow=alaw EXTENSIONS.CONF contents: TRUNK=Zap/g1 TRUNK=Zap/g2 [tutorial] exten = s,1,Dial(SIP/350,30) exten = s,n,Voicemail(350) exten = s,n,Hangup exten = 300,1,Dial(SIP/300,15) exten = 300,2,Voicemail(u300) exten = 300,3,Voicemail(b300) exten = 300,4,Hangup What is the solution? Please tell me. Looking forward to your response. Thank you. Regards, Chandra. __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with call receiving (Asterisk+PSTN+Digium TDM04B)
First zaptel.conf fxsks=1-4 zapata.conf channel = 1-4 extensions.conf [tutorial] exten = s,1,Dial(SIP/350,30) - do You have SIP/350 ? ther is onlu 300 in sip.conf Hi Friends, I am Chandra from India. Thank you for your cooperation and for clear my doubts. Now, I have installed Digium TDM04B card in my Asterisk server and configured. I have one landline number from PSTN. Now, I have connected that PSTN cable to my TDM04B first port. When I am making calls from outside to my PSTN number, sometimes Asterisk receving that call and sometimes, its not receiving. Why? Here I am giving my configuration of my files. ZAPTEL.CONF contents: loadzone = us defaultzone=us fxsks=1,2,3,4 ZAPATA.CONF contents: [channels] context=tutorial signalling=fxs_ks busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes cancallforward=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callerid=asreceived language=en usecallerid=yes echocancel=yes transfer=yes immediate=no group=1 channel = 1 SIP.CONF contents: [300] type=friend username=300 secret=server callerid="Server" host=dynamic context=tutorial [general] port=5060 bindaddr=0.0.0.0 context=default disallow=all allow=gsm allow=ulaw allow=alaw EXTENSIONS.CONF contents: TRUNK=Zap/g1 TRUNK=Zap/g2 [tutorial] exten = s,1,Dial(SIP/350,30) exten = s,n,Voicemail(350) exten = s,n,Hangup exten = 300,1,Dial(SIP/300,15) exten = 300,2,Voicemail(u300) exten = 300,3,Voicemail(b300) exten = 300,4,Hangup What is the solution? Please tell me. Looking forward to your response. Thank you. Regards, Chandra. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dropping calls in the middle of conversation
well, first thing, turn on debug logging in logger.conf (edit the messages line so that it includes the word debug, the file has examples to help you). then, after doing a logger reload, you will be getting quite a bit of logging...the next time a call drops, note as much about it as you can (the destination number works great for me) then search /var/log/asterisk/messages for that number. look at the information and see if any of it makes sense. we are having dropped calls with our telco (SBC) and 99% of them are on the telco's side, beyond the CO. all i do now is look for a didn't receive frame from Zap/XX - that tells me that the telco stopped transmitting, so asterisk hangs up the call. every one of the calls that i've had SBC trace has been terminated by the far side... check out your debug log and post back if you still need assistance. wes On 7/27/06, Giannis Margaritis [EMAIL PROTECTED] wrote: Hi all, I'm having major trouble with a simple asterisk installation dropping calls in the middle of the conversation. I recently upgraded from asterisk-1.2.3 and zaptel-1.2.2 to asterisk 1.2.10 and zaptel-1.2.7, but to no avail. The machine is equiped with a TDM40B and a TDM22B and has an MSI motherboard with an intel 915G chipset and a SATA hard disk 80 GB. It's running Fedora core 3 Linux. The situation appears with no obvious reason, the CLI shows nothing more than the zaptel channel hanging up. How should i go debugging this mess? Giannis Margaritis ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Developing VoIP with Asterisk
Hi Group!Hi Wagner!Thanks for the interest. I'm from Colombia and I'm trying to develop VoIP as you know on *. So thanks again for the offering in Brazil, althought you can help me with some idea by this way. To make the call I'm using SJphone (softphones) to make the tests. I'm not using IP phones because we don't have a lot of investment as a said before.This is my [general]sip.conf format: I omitted other parts which were on comments because it is example from the web site [general]context=default;allowguest=no ;realm=mydomain.tld bindport=5060bindaddr=0.0.0.0srvlookup=yes ;domain=mydomain.tld ;** Cambio de lineasdisallow=all;allow=g729allow=gsmallow=ulawjitterbuffer=yesmaxjitterbuffer=800;allow=ilbc;musicclass=default;language=en;relaxdtmf=yesrtptimeout=60 ;rtpholdtimeout=300;trustrpid = no;sendrpid = yes;progressinband=never ;useragent=Asterisk PBX;promiscredir = no ;usereqphone = no ;*** Cambio de lineas DTMFMODE estaba en comentarios dtmfmode = rfc2833 ;compactheaders = yes ;sipdebug = yes ;subscribecontext = default ;notifyringing = yes ; Usuario 1 [usuario1] type=friendhost=dynamicdtmfmode=rfc2833username=usuario1secret=usuario1; Usuario 2 [usuario2]type=friendhost=dynamicdtmfmode=rfc2833 username=usuario2secret=usuario2Thanks again for the interest and if you have and idea I would apreciate a lot!Carlos Bernat2006/7/27, Wagner Nunes [EMAIL PROTECTED]: Hi Carlos!!!Let me ask one thing... ... r u brazilian???Becouse I work with * projects and if u r in brazil maybe i can help u. But about your problem, What are u using to call thru *? IP Phone, softphone? What is your sip.conf settings? Carlos Alberto Bernat Orozco [EMAIL PROTECTED] escreveu: Hi Group!Still I'm concern about my problem with echo on the voice and I want to ask some advice to developing VoIP. Maybe I'm very ambiciuos or maybe not because I want to give VoIP to near 500 users. We got an small ISP and we have the project to give telephony (for now) to our users between them. Our resources are limited and I have installed * as a hope to give this service to our users. We have a good network (with small problems) but I believed that is possible to give this service. Our HFC network is very well calibrated and works fine. The users have cable modems to connect to the internet and we give private adresses to some users. I'm searching for someone who has the same problem in the past with similar things, to know how solve it and if is possible to give VoIP calls with a server with a public address and the softphones (for the costs) with extensions registered on our * box. I configured * four months ago and between two extensions and works very well and but later I did the same test on this week and unfortunaly the voice goes out with echo. So I have the feeling that maybe there's something wrong with the codecs and wich codecs do I need to give the service. Thanks for any help you can give meCarlos Bernat___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Você quer respostas para suas perguntas? Ou você sabe muito e quer compartilhar seu conhecimento? Experimente o Yahoo! Respostas! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys SPA-3102
Has anyone used the new 3102? If so, does it work correctly? I heard lots of horror stories about the SPA-3000 causing terrible echo, picking up voice tones as DTMF, etc, so Im a little hesitant to buy. Thanks, Wes Baehr -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.10.4/401 - Release Date: 7/26/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Mobile SIP Client
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shad Mortazavi Sent: Thursday, July 27, 2006 4:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Mobile SIP Client Dear All, I'm looking for a mobile SIP client to use with Asterisk. Has anyone got experience in this area and can you advise me of a product? Do you have any preference for the mobile operating system? Windows Mobile 5 is an obvious choice: lots of handhelds already have it, it is a well-understood product with mature development tools available, and it is a reasonably open system. On the other hand, linux ports to PDAs are common these days, and, depending on your available developer talent, may present a more economical product development path. If you are contemplating a WiFi airlink, be sure to do your due diligence on the topic of RF propagation--early reports of SIP over WiFi are sobering. If you are thinking of multi-airlinks, either WiFi Bluetooth or WiFi cellular, it is best to reference what the handheld manufacturers are doing in this complex area. Take a look at PCTEL's Roaming Client work for guidance. For a good summary of the issues, look at the table of contents of an industry report on mobile handset convergence: http://www.disruptive-analysis.com/sip_and_ims_handsets_toc.htm HTH, James Hanlon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Error when calling
Dear Anthony, I believe you where right the dial plan seems to have been missing the TRUNK= statement and I found one in the file extensions.conf but not the correct group I configured so I changed it and will test again. Thx MAG "Mohamed A. Gombolaty" wrote: Dear Anthony, The dial plan is currently very simple it should pick up any call and send it to a sip phone registered, you can see the context below named zap-in is what I am using, it is only that and nothing more, is there something extra I have to add to dial plan or to that context ? Thx MAG Anthony Rodgers wrote: This looks like a dialplan problem - do you have a match for 0109687348 in the zap-in context in your dialplan? A. On 26-Jul-06, at 2:40 PM, Mohamed A. Gombolaty wrote: > Dear All, > I have a strange problem in recieving calls on the pri the zaptel > is green and everything seems very well, but when a call comes I > can see the call along with the caller ID but then I get this > strange message which make the call hungup: > > > error msg: 'zap-in' from '0109687348' does not exist. Rejecting > call on channel 0/18, span 1. > > the PRI is an E1 and I have the following configuration for > extensions.conf > > [zap-in] > exten => s,1,Answer > exten => s,2,Dial(sip/100) > exten => s,3,Hungup > > as for the zapata.conf it is as follow: > > [channels] > language=en > switchtype=euroisdn > signalling=pri_cpe > context=zap-in > group=0 > channel=>1-15,17-31 > > I don't know what the problem is or where to look, I will > appreciate it if someone can help me out? > > Thx > MAG > > -- Thx MAG > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] bugs.digium.com
I opened bug #0007490 the other day. The issue was that when you do a 'sip debug' on the Asterisk console, there was no way to have this output go _only_ to the messages file. Someone with the id of 'russell' in his infinite wisdom has deemed that this isn't a bug, closed it, and given me -2 karma points. WTF??? It clearly is a bug, or at the VERY least, a limitation that needs to be fixed. So why the hell did he give me -2 karma points and say 'not actually a bug'. Fine... so how do you file an enhancement request then? If there's no way to file an enhancement request, then this is the most appropriate place to file this. Its damn irritating not being able to have 'sip debug' output go to a file only, and this is what the options in logger.conf implyyou should be able to do, which is another reason Idon't understand why he took this irrational action. In a PRODUCTION environment, you can't be running a sip debug to your console. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bugs.digium.com
On Thursday 27 July 2006 10:32, Douglas Garstang wrote: It clearly is a bug, or at the VERY least, a limitation that needs to be fixed. So why the hell did he give me -2 karma points and say 'not actually a bug'. Fine... so how do you file an enhancement request then? If there's no way to file an enhancement request, then this is the most appropriate place to file this. When I report a bug, I can say it's for a Feature Request. Perhaps that's what you should have done? Its damn irritating not being able to have 'sip debug' output go to a file only, and this is what the options in logger.conf imply you should be able to do, which is another reason I don't understand why he took this irrational action. It's perfectly rational. You posted a bug that is at best a feature request. That's where the -2 came from. I agree with you in the sense that it should not have been closed but simply readdressed, but that's not my call. In a PRODUCTION environment, you can't be running a sip debug to your console. In a PRODUCTION environment you have all of these issues worked out in your test lab before deploying to production. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bugs.digium.com
I hate to say this but you might just have hit a 'reap what you sow' moment - you don't hesitate to trash Asterisk on this mailing list when you can't make it do what you think it should do, and just maybe this affects how the developers treat requests from you on the bug tracker? Just a thought. Peter On 27/07/06, Douglas Garstang [EMAIL PROTECTED] wrote: I opened bug #0007490 the other day. The issue was that when you do a 'sip debug' on the Asterisk console, there was no way to have this output go _only_ to the messages file. Someone with the id of 'russell' in his infinite wisdom has deemed that this isn't a bug, closed it, and given me -2 karma points. WTF??? It clearly is a bug, or at the VERY least, a limitation that needs to be fixed. So why the hell did he give me -2 karma points and say 'not actually a bug'. Fine... so how do you file an enhancement request then? If there's no way to file an enhancement request, then this is the most appropriate place to file this. Its damn irritating not being able to have 'sip debug' output go to a file only, and this is what the options in logger.conf imply you should be able to do, which is another reason I don't understand why he took this irrational action. In a PRODUCTION environment, you can't be running a sip debug to your console. Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bugs.digium.com
Dear All, I just wanted to comment on this point of the discussion: > In a PRODUCTION environment, you can't be running a sip debug to your > console. In a PRODUCTION environment you have all of these issues worked out in your test lab before deploying to production. I do agree with Douglas that having a way to log the debug of sip to a file would be a great option available to use in production, you cannot test a problem occuring to a production system in the lab or even expect problems before going into production to resolve them in the lab, I believe the russel didn't understand well what the request was. But I do hope you can file the bug and really make it obvious that it's a feature request, and I believe someone will take care of it. Thx MAG Andrew Kohlsmith wrote: On Thursday 27 July 2006 10:32, Douglas Garstang wrote: > It clearly is a bug, or at the VERY least, a limitation that needs to be > fixed. So why the hell did he give me -2 karma points and say 'not actually > a bug'. Fine... so how do you file an enhancement request then? If there's > no way to file an enhancement request, then this is the most appropriate > place to file this. When I report a bug, I can say it's for a "Feature Request". Perhaps that's what you should have done? > Its damn irritating not being able to have 'sip debug' output go to a file > only, and this is what the options in logger.conf imply you should be able > to do, which is another reason I don't understand why he took this > irrational action. It's perfectly rational. You posted a bug that is at best a feature request. That's where the -2 came from. I agree with you in the sense that it should not have been closed but simply readdressed, but that's not my call. > In a PRODUCTION environment, you can't be running a sip debug to your > console. In a PRODUCTION environment you have all of these issues worked out in your test lab before deploying to production. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP client with video???
Hello to all can someone recommend me a nice SIP client with video for windows?? I tried X-Lite 3.0 but it's a lousy piece of software. Does someone knows about a better software? Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] bugs.digium.com
-Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: Thursday, July 27, 2006 8:48 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] bugs.digium.com On Thursday 27 July 2006 10:32, Douglas Garstang wrote: It clearly is a bug, or at the VERY least, a limitation that needs to be fixed. So why the hell did he give me -2 karma points and say 'not actually a bug'. Fine... so how do you file an enhancement request then? If there's no way to file an enhancement request, then this is the most appropriate place to file this. When I report a bug, I can say it's for a Feature Request. Perhaps that's what you should have done? Its damn irritating not being able to have 'sip debug' output go to a file only, and this is what the options in logger.conf imply you should be able to do, which is another reason I don't understand why he took this irrational action. It's perfectly rational. You posted a bug that is at best a feature request. That's where the -2 came from. I agree with you in the sense that it should not have been closed but simply readdressed, but that's not my call. I really don't believe that it's a feature request. I belive it's a bug. By putting 'debug' against messages, and not against console, any sane person would think that debug (ie as a result of typing 'sip debug' would go to the messages file, and not to the console. In a PRODUCTION environment, you can't be running a sip debug to your console. In a PRODUCTION environment you have all of these issues worked out in your test lab before deploying to production. In a PRODUCTION environment, you will encounter issues. It happens. That's life. You need to be able to debug these problems. You can't possibly think that when you roll this out from dev to production, that there will be no issues. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Rxfax and squashed TIFF
Garth van Sittert wrote: Hi All Just wondering if anyone knows of a solution to the squashed tiff problem with spandsp (or rather Windows Image Viewer) other than converting to a PDF. I find the PDF image quality is not nearly as good as the original TIFF. Apparently the Windows Image Viewer doesn't understand the metric units in the TIFF header. Convert them to imperial units? Kind Regards Garth You must be using an old version of spandsp. Newer ones use imperial measurements to improve compatibility with broken viewers. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mobile SIP Client
Thank you for the information. I'm specifically looking for a Windows 5.0 Mobile SIP agent for a Qtek 9000. Many Thanks Shad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP client with video???
What's wrong with X-Lite 3.0? I haven't had any issues with it and find it to be one of the best SIP video software choices, and it's free.On 7/27/06, Joao Pereira [EMAIL PROTECTED] wrote:Hello to all can someone recommend me a nice SIP client with video for windows??I tried X-Lite 3.0 but it's a lousy piece of software.Does someone knows about a better software?RegardsJoao Pereira ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip realtime
On Thursday 27 July 2006 15:07, Andrea Spadaccini wrote: Ciao Benchev, Also register= can be done only from a .conf file. Well, I'm experimenting right now with this, and I can tell you that register = works even with static realtime. Not even, it *must* work because if one uses realtime static, the equivalent file in /etc/asterisk i.e. sip.conf, should be deleted. Ciao, Benchev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recommend hard phone which supports IAX2?
Michael Graves wrote: On Tue, 25 Jul 2006 19:43:58 +0100, Tim Panton wrote: On 25 Jul 2006, at 16:23, Stephen Bosch wrote: What are the best IAX2 hard phones? I've got a couple of IAX hardphones, with PA168, they are useable, but only just. They are hard to hang up (which is a design problem) and a pain to get transfer working (which is a software problem). Much as I love IAX, I advise you to buy a decent SIP phone (SNOM?). At home I have a SIP phone and an nslu2 running asterisk, just to act as a protocol converter, but any old 486 or PII will do the trick. I echo this sentiment. Except that I'd recommend Astlinux on a WRAP or Soekris board. Small, low power,fanless, boots from CF or USB key and able to transcode between G.711 and G.729a. Astlinux rocks! I use Soekris units with OpenBSD for VPN edge devices. I take it you're running Astlinux on a Soekris? That's worked well for you? How's the performance? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Mobile SIP Client
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shad Mortazavi Sent: Thursday, July 27, 2006 10:20 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Mobile SIP Client Thank you for the information. I'm specifically looking for a Windows 5.0 Mobile SIP agent for a Qtek 9000. The PCTEL client looks OK: http://mobilitysolutions.pctel.com/product_overview_detail.cgi?id_num=10813 Counterpath is supposed to have a WM5 port soon. Or see Wikipedia on SIP clients. If you find yourself setting up test scenarios for multiple WM5 devices, performing either voice interactions or simultaneous voice/data, contact me off-list for information on automating the process. Regards, James Hanlon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfers - No ringback or moh
Hi Mike, Hi all, really works. ;-) But that can not be the solution for the future? :-) Can it? I think there should be an ANSWER() implimented in the Transfer function to prevent this problem ... Or does anybody have other ideas? greetings, Martin - Original Message - From: Mike Dawson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 26, 2006 4:32 PM Subject: Re: [asterisk-users] Transfers - No ringback or moh I get round this bug by replacing: exten = X,1,Dial(sip/blah) with: exten = X,1,Answer exten = X,n,Dial(sip/blah) It means the call is in an answered state before it starts ringing but it doesn't seem to cause any major problems. Mike Martin Schrott - Thinking-Systems wrote: Hi all, I cannot exactly reproduce your problems, but I can tell you, what problem we have on this topic: a calles b. b takes the call and can speak to a. b sets up a attendend transfer (via the softkey configured in asterisk) to c and hears ringing. a hears music on hold. b hears ringing if c answeres and b hanges up, everything is fine. now the problem: if b hangs up, before c has answered (during ringing) a will loose the connection and also be hanged up. I think this should not happen! The transfer should automatically be changed to blind and a should get the ringing played back instead of b. Hope, you can understand my problem and may have any ideas or thoughts. Greetings and Thanks, Martin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Mobile SIP Client
Hi Shad, If you haven't committed to your handsets already, you might want to wait it out a few weeks for the HTC Hermes. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Shad Mortazavi Sent: Thursday, 27 July 2006 11:20 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Mobile SIP Client Thank you for the information. I'm specifically looking for a Windows 5.0 Mobile SIP agent for a Qtek 9000. Many Thanks Shad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: dropping calls in the middle of conversation
I'm having major trouble with a simple asterisk installation dropping calls in the middle of the conversation. I was having similar issues when we installed a new non-pri T1, one of the problems had to do with the wiring job that was done, but the major problem seemed to be related to IRQ sharing, I disabled EVERYTHING I could on the motherboard, including USB interfaces (which seemed to always be sharing IRQs with one of the Digium cards). After turning all motherboard peripherals off, the number of dropped calls fell to about 2/week, before doing this we were dropping lots and lots of calls every day. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF relay
- Original Message - From: Jason Kim [mailto:[EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thu, 27 Jul 2006 01:51:19 -0300 Subject: [asterisk-users] DTMF relay Hi, My environment is ITSP---Asterisk--SipPhone. I want to send dtmf from SipPhone to ISTP using 'info' or 'rfc2833'. Is this possible? It sure is - your ITSP just has to support it (most do support RFC2833 though so you should be fine). Each side can even use different DTMF modes, Asterisk will automatically take care of it. Just make sure dtmfmode in sip.conf for each entry is set to the one you want. Thanks. Jason. Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Message waiting question...
- Original Message - From: Jean-Yves Avenard [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Thu, 27 Jul 2006 02:07:56 -0300 Subject: Re: [asterisk-users] Message waiting question... Hi On 7/27/06, Joshua Colp [EMAIL PROTECTED] wrote: chan_sip requests the count fairly frequently, dunno how much traffic it would actually generate though. Well I took the very easy route. Every minute I do a rsync between server2 and server1 of the INBOX directory I want to check. I also only transfer the .txt file so it never needs to transfer more than 500 bytes max every minute. Having just the .txt file is sufficient for Asterisk to tell the SPA3000 that there's a message waiting. And best of all: it works :) As long as it works - that's great! As a side question, is there a way to force asterisk to set specific group permission on the file generated for the voicemail? I found some patches for earlier version of Asterisk and at one stage that it made its way into asterisk trunk I can't find any documentation about how to configure it though. I don't believe there's anything configurable but if you open app_voicemail.c there's two declarations, VOICEMAIL_DIR_MODE and VOICEMAIL_FILE_MODE which set the permissions. DIR mode is at 0770 right now and FILE mode is at 0660. Thanks JY Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Woes
Hi all, Hiya. After some more thought and investigation, I think the following is definitely my problem: - Looking for 10 in Outgoing Reliably Transmitting (NAT): SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP - Indeed. Does anyone know how I can resolve this ? - Incoming calls work fine, internal calls work fine (so to talking clock etc) but outgoing do not. My configuration can be found here: http://files.davehope.co.uk/asterisk-problem/ exten = _X,1,Dial(SIP/[EMAIL PROTECTED],30,trg) exten = _X,2,Hangup Those two lines will only be matched if a person only dials 1 digit extensions from 0 through 9, what you probably meant is: exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],30,trg) exten = _X.,2,Hangup Notice the . after the X? It means match any extension starting with 0 through 9, of any length. If anyone would be so kind as to shed some insight into the matter it'd be greatly appreciated!, Kind Regards, Dave Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] odd sound between SIP IAX clients
Pure guess without looking at the trace yet, its likely to be a timing issue (involving the codec translations) as opposed to the codec itself. I've had good luck running g726 on iax links. Joseph Love wrote: Well, you win, it's definitely codec related. Switching to ulaw causes this issue to go away. I find it rather peculiar that the issue crops up in the first place. If anyone is interested in packet traces of the problems I'm encountering with GSM, I can manage to arrange for those. Thanks, -Joe On Jul 25, 2006, at 9:50 PM, Rich Adamson wrote: Joseph Love wrote: The issue which occurs is that the audio from the SIP client to the IAX client will spend most of it's time sounded very robotic, and garbled. It is possible, although very difficult to understand someone who is on the SIP phone. I have asterisk 1.2.10 configured with realtime with both IAX and SIP clients. The SIP clients include a Grandstream gxp2000 hard phone, and Counterpath's X-Lite 3 (for windows) softphone. The IAX clients tested include idefisk (both windows mac), JakenIAX, and LoudHush. GSM is the preferred codec of both IAX SIP clients, and is indeed the codec being used in all tests. Audio from the IAX to the SIP client does not experience any issues. SIP to SIP (and presumably, although untested, IAX to IAX) communication does not experience any issues. We also have a T1 card through which many calls have been placed, both from the IAX and SIP phones, without any audio issues occurring, in either case. If it weren't for that there have been multiple clients tested to verify this robotic sound, I would cough it up to it being a incompatability between the particular clients, but this occurs on all SIP-IAX communication that has been tried. I'm running out of options as SIP-IAX intercommunication is kinda expected (and necessary for me), and out of good softphones for the mac, as most of the mac-compatible softphones are IAX2-based. Please let me know what additional information is needed to help me debug this problem. Can you try different codecs just to rule out any issues with that? E.g., if both devices use ulaw, do you still have the same problem? I've used both iaxcomm and x-lite to communicate with cisco, polycom, grandsteam, etc, without that type of problem. Is it possible to obtain an ethereal trace of both the iax and sip/rtp streams in the same trace? If so, a couple hundred packets should be more then enough to see what's going on. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] bugs.digium.com
On Jul 27, 2006, at 7:32 AM, Douglas Garstang wrote: I opened bug #0007490 the other day. The issue was that when you do a 'sip debug' on the Asterisk console, there was no way to have this output go _only_ to the messages file. Someone with the id of 'russell' in his infinite wisdom has deemed that this isn't a bug, closed it, and given me -2 karma points. WTF???This sounds like a problem to me for the following reasons.1) It does seem like a bug, and "Russell" didn't bother explaining why it isn't. This is a basic feature that should work (it seems).2) Some people here seem to think it's OK if Douglas's personal attitude make it OK for his bug reports to be ignored. This is just bad baseball.That said, Douglas, you do seem to go out of your way to generate negative Karma.my 2c US,Marty___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] OFF-TRACK: Is VOIP -PSTN integration legal inChina
Dashy wrote- Hi Dan, Thanks for the reply. I was just searching the MII china site to confirm this. But could not find any information on the same. They just kept talking about issuing VOIP licences to service providers But no focus on whether an enterprise can do it for itself or not. But just one point, Do the PSTN service provider made some checks on what have you installed? Before deploying any equipment I had meetings with the local telco in three major cities in China. In each meeting I expressed our interest to connect Cisco VoIP technologies, but would follow the law and purchase an approved PBX if required. In each meeting the telco reps were very clear. We want to sell service, the termination doesn't matter to use and the law permits it. There is/was a gray area if the equipment was used to bypass toll charges NOT related to normal company business. So using our infrastructure to offer non-employees a way around the local telco would cause problems, but that is not something I wanted to support anyways. I too agre with you about India. Sad..isn't it? Yes an no. It made my deployment not meet my own policies and standards, but I simply refused local PSTN service. Our employees can use their IP phones to communicate with their international peers, and have to use either their cell phones or one of four analog lines in common areas for in country calls. Our environment permitted such a restriction. So BNSL/VNSL lost out on service revenues, and I lost no sleep over the decision. Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] alcatel ip touch 4068 ... sip?
2006/7/27, Cesc [EMAIL PROTECTED]: Hi,Quickie ... is the alcatel ip touch 4068 (or any other in that series)sip enabled?If not, does alcatel have a sip-enabled phone?CescNo (for both questions), as SIP is seen as a low end protocol, yet unable to transport high end features of ip touch phones. Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: dropping calls in the middle of conversation
Have you tried setting: -- Faxdetect=no Busydetect=no Callprogress=no Busycount=8 -- In zapata.conf? John Coleman - IT Specialist SunWest Education Credit Union http://www.swecu.com well, first thing, turn on debug logging in logger.conf (edit the messages line so that it includes the word debug, the file has examples to help you). then, after doing a logger reload, you will be getting quite a bit of logging...the next time a call drops, note as much about it as you can (the destination number works great for me) then search /var/log/asterisk/messages for that number. look at the information and see if any of it makes sense. we are having dropped calls with our telco (SBC) and 99% of them are on the telco's side, beyond the CO. all i do now is look for a didn't receive frame from Zap/XX - that tells me that the telco stopped transmitting, so asterisk hangs up the call. every one of the calls that i've had SBC trace has been terminated by the far side... check out your debug log and post back if you still need assistance. wes On 7/27/06, Giannis Margaritis [EMAIL PROTECTED] wrote: Hi all, I'm having major trouble with a simple asterisk installation dropping calls in the middle of the conversation. I recently upgraded from asterisk-1.2.3 and zaptel-1.2.2 to asterisk 1.2.10 and zaptel-1.2.7, but to no avail. The machine is equiped with a TDM40B and a TDM22B and has an MSI motherboard with an intel 915G chipset and a SATA hard disk 80 GB. It's running Fedora core 3 Linux. The situation appears with no obvious reason, the CLI shows nothing more than the zaptel channel hanging up. How should i go debugging this mess? Giannis Margaritis ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reload of wct4xxp without restarting of Asterisk?
On Thu, 2006-07-27 at 09:40 +0200, [EMAIL PROTECTED] wrote: As i writed, i do an unload of chan_zap.so but i can't unload the module wct4xxp, is this possible? Well, you actually said that the error was for chan_zap.so. :) Anyway, you should never need to unload the wct4xxp driver. If you need to make configuration changes, then you just re-run ztcfg. -- Russell Bryant Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Woes
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Joshua Colp wrote: Hi all, Hiya. After some more thought and investigation, I think the following is definitely my problem: - Looking for 10 in Outgoing Reliably Transmitting (NAT): SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP - Indeed. Does anyone know how I can resolve this ? - Incoming calls work fine, internal calls work fine (so to talking clock etc) but outgoing do not. My configuration can be found here: http://files.davehope.co.uk/asterisk-problem/ exten = _X,1,Dial(SIP/[EMAIL PROTECTED],30,trg) exten = _X,2,Hangup Those two lines will only be matched if a person only dials 1 digit extensions from 0 through 9, what you probably meant is: exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],30,trg) exten = _X.,2,Hangup Notice the . after the X? It means match any extension starting with 0 through 9, of any length. Thanks for the suggestion, I added that in and now get: Jul 23 16:57:31 WARNING[4114]: pbx.c:1292 pbx_extension_helper: No application 'Dial' for extension (Outgoing, 10, 1) Reliably Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.11:5064;branch=z9hG4bK7a6c25f1-041c-db11-82b2-000fea3f84d4 And, to make sure I didn't make a type in my dialplan: exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],30,trg) exten = _X.,2,Hangup Any thoughts ? Dave If anyone would be so kind as to shed some insight into the matter it'd be greatly appreciated!, Kind Regards, Dave Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.4 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEyPvajdL3ZT1KDlERAsRVAJ0Yc6tHgshPiihTenkd2mu+bbl4eACgoPF+ RJZjZoSB3xSXL1QsfpyBT5w= =/AiD -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Woes
- Original Message - From: Dave Hope [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Thu, 27 Jul 2006 14:46:02 -0300 Subject: Re: [asterisk-users] SIP Woes Thanks for the suggestion, I added that in and now get: Jul 23 16:57:31 WARNING[4114]: pbx.c:1292 pbx_extension_helper: No application 'Dial' for extension (Outgoing, 10, 1) Reliably Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.11:5064;branch=z9hG4bK7a6c25f1-041c-db11-82b2-000fea3f84d4 And, to make sure I didn't make a type in my dialplan: exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],30,trg) exten = _X.,2,Hangup Any thoughts ? app_dial.so is not loaded, so the Dial dialplan application does not exist. You can load it from the CLI by doing load app_dial.so or explicitly putting it in your /etc/asterisk/modules.conf to be loaded when Asterisk starts. Dave Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Goldmine SIP client/softphone questions continued:
Hi all, still trying to debug this Goldmine CRM softphone, it does appear that the client is being authenticated, but the server is replying with this message (in the /var/log/asterisk/full file) DEBUG[4307] chan_sip.c: SIP message could not be handled, bad request: [EMAIL PROTECTED] Does that mean anything to anyone? Thx again Dan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Woes
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Joshua Colp wrote: - Original Message - From: Dave Hope [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Thu, 27 Jul 2006 14:46:02 -0300 Subject: Re: [asterisk-users] SIP Woes Thanks for the suggestion, I added that in and now get: Jul 23 16:57:31 WARNING[4114]: pbx.c:1292 pbx_extension_helper: No application 'Dial' for extension (Outgoing, 10, 1) Reliably Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 192.168.1.11:5064;branch=z9hG4bK7a6c25f1-041c-db11-82b2-000fea3f84d4 And, to make sure I didn't make a type in my dialplan: exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],30,trg) exten = _X.,2,Hangup Any thoughts ? app_dial.so is not loaded, so the Dial dialplan application does not exist. You can load it from the CLI by doing load app_dial.so or explicitly putting it in your /etc/asterisk/modules.conf to be loaded when Asterisk starts. Wow. How did I miss that one :) Thanks for your help! Dave Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.4 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEyQSzjdL3ZT1KDlERAseBAJ48gG3LVAyrRDZQWv/5e7Bc0DdOQgCePBp0 dy6clOj15wrVt+nGVcVLEB4= =ywbJ -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP phone w/ 'modem/data' port?
Hi all, has anyone EVER seen a SIP phone that has a 'data port', like most business phones do? We use this in our office to connect analog wireless headsets to the desk phones (plantronics ct12s) and need to continue using these (as we have a significant amount of $$$ invested in them). I've already setup FXS ports for phones can use analog desk phones to accomplish this, but the Aastra ADSI phones that I've tried all seem to have some kind of problem w/asterisk or the channel bank (most notably an inability to display caller id when sent from asterisk via channel bank, although other analog CID phones CAN display incoming info). Does a sip phone exist with an analog port to connect a normal phone/modem to? Or does anyone know of an inexpensive ADSI phone that'll actually work w/asterisk? Thanks as always. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Dial Tone
Hi, i'm having problems with DTMF, the problems are with established connections and some IVRS. When i call to other number which has an IVR, some digits doesn't work. I digit a long number (required by the IVR, at least a 10 digit number) and it doesn't work. I think it's about DTMF signalling, i've all my extensions with RFC2833 mode, i've an LinkSys PAP-2 and a Polycom 301, allowed codec is ulaw. If you need more information, pleas feel free to ask :) Cheers, Santiago ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Goldmine SIP client/softphone questions continued: (Dan Elder)
Hi all, still trying to debug this Goldmine CRM softphone, it does appear that the client is being authenticated, but the server is replying with this message (in the /var/log/asterisk/full file) DEBUG[4307] chan_sip.c: SIP message could not be handled, bad request: [EMAIL PROTECTED] Just turned up the level of debug got these in the logs: Jul 27 12:11:11 DEBUG[4307] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag 3b77d0 Our tag: as2905cb31 Jul 27 12:11:11 DEBUG[4307] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag 3b77d0 Our tag: as2905cb31 Jul 27 12:11:16 DEBUG[4307] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag 3b77d0 Our tag: as2905cb31 Jul 27 12:11:18 DEBUG[4307] chan_sip.c: = Found Their Call ID: [EMAIL PROTECTED] Their Tag 3b77d0 Our tag: as2905cb31 Jul 27 12:11:18 DEBUG[4307] chan_sip.c: SIP message could not be handled, bad request: [EMAIL PROTECTED] Jul 27 12:11:20 DEBUG[4307] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag 3b77d0 Our tag: as2905cb31 Jul 27 12:11:21 DEBUG[4307] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag 3b77d0 Our tag: as2905cb31 Jul 27 12:11:22 DEBUG[4307] chan_sip.c: = No match Their Call ID: [EMAIL PROTECTED] Their Tag 3b77d0 Our tag: as2905cb31 Jul 27 12:11:25 DEBUG[4307] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Is this Frontranges softphone sending things incorrectly? Or any other ideas what the above might mean? Thanks again...sorry for all the traffic 2day.. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone tried vitelity?
I was just wondering if anyone out there has tried vitelity for VoIP service If you did what is your story with how good/bad they are? Thanks! Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Detecting voicemail from CO on FXO port and passing to H.323 phone. Possible?
The subject pretty much describes what I need to do. Basically, I want to be able to detect that there is voicemail waiting at the CO on an FXO port and somehow flash the message waiting light on an H.323 phone (or any other type of phone) I my case, the CO is actually a legacy POTS based PBX and I've just plugged the FXS station ports into my FXO ports on asterisk. The PBX is able to set the MWI lights on my POTS phones so I know it has the capability to tickle the FXO port on my asterisk box. What I don't know is how to detect that in asterisk and then tell the H.323 phone to flash its light. Any clues would be appreciated. Bob. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone tried vitelity?
On Thu, 2006-07-27 at 15:36 -0400, Curt Shaffer wrote: I was just wondering if anyone out there has tried vitelity for VoIP service If you did what is your story with how good/bad they are? I have just recently made the switch from Sixtel to them (because Vitelity bought Sixtel). For the moment everything seems to be working and at least their support is a lot more responsive that sixtel used to be. -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone tried vitelity?
I have used exgn for several months and after the merger I have had now problems.On 7/27/06, Curt Shaffer [EMAIL PROTECTED] wrote: I was just wondering if anyone out there has tried vitelity for VoIP service If you did what is your story with how good/bad they are? Thanks! Curt ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 Connection fails over time...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hey all, I have a x86 Pentium D asterisk system with two Digium 400's in it. I am establishing a IAX2 Connection to another Asterisk system running on a Solaris server. When a call is placed between the two systems, everything seems fine for a variable period of time, then for some reason beyond what my diagnostics has found, the call begins to lag, and both asterisk's servers report LAG. Network wise, the systems have a 4-10 msec ping time. Once the call is terminated, everything returns to normal, and the call can be reconnected. Until the call is terminated, no other calls can be setup with that host. Both systems are running 1.2.x. We are using the GSM codec for the calls. Any ideas on what we should check? Stu - -- Randomly Generated Fortune Tag: There is a great discovery still to be made in Literature: that of paying literary men by the quantity they do NOT write. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.3 (GNU/Linux) iD8DBQFEySV/K69Y+xPZrWYRAg2fAKCPy4inQdicqTrwvoW+VQMRS/vlIQCfVSLT OC0G/AfRej+vzfdvw/tgfbY= =Ck7n -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] gxp-2000 configure line appearances
Can anyone tell me how to configure the grandstream gxp-2000 for 4 line apearances. I have the the sample conf from the website and the phone is getting its config from my TFTP server. But it does not have any info for the other line apearance butons The real thing that would help is a complete list of the configurable comands on the latest firmware so I can create the config file. # Admin password for web interface P2 = # SIP Server P47 = # Outbound Proxy P48 = # SIP User ID P35 = Any help would be great. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Getting no Audio with G729
Hello, Recently I purchased g729 codec and installed in Tribox 1.1(upgraded 1.1.1)/ Asterisk. I have pointed a DID from my carrier via SIP through g729 to asterisk. Problem is I am not getting any audio even though I am getting DTMF in asterisk. I am trying to run A2billing with asterisks. Configuration of carrier is asterisk is: [abc] allow=g729 context=c-DID dtmfmode=auto host=xxx.xxx.xxx.xxx insecure=very sendrpid=yes type=friend echo=no Any suggestions ? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager interface
On 27 Jul 2006, at 11:47, Lee Archer wrote: This has probably been discussed before but I need to do a screen pop and I'm looking for ways to do it. I am assuming I need to use the manager interface, which is ok cos I'm using that for calling out but I'm not quite what to pick up on. There a number of ways to do this: 1) run an application on each workstation which speaks the manager protocol and pops a screen as needed. This doesn't scale easily to large numbers, you need to install an application on each workstation and need some sort of manager proxy as asterisk does not like many manager connections. 2) run an IM client on each workstation and have a central server that talks the manager protocol to asterisk, sending messages to IM clients when new calls come in. 3) have each user point their webbrowser at a web server which talks the manager protocol to asterisk and have the webpage poll the server (using AJAX) 4) embed a softphone in your application (or web page) and send calls to it. Configure the softphone to pop the screen when a call comes in. We do 4) . which you chose depends on your needs/skills. Tim. Regards Lee ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Anyone tried vitelity?
I am with Vitelity now, they just completed merging with EXGN which i was signed up with. I also signed up with them with some of my clients. So far (last 4months) no issues what so ever. Great service, and Customer support is timely and very knowledgeable. Terrelle From: Curt Shaffer [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: [asterisk-users] Anyone tried vitelity? Date: Thu, 27 Jul 2006 15:36:11 -0400 I was just wondering if anyone out there has tried vitelity for VoIP service If you did what is your story with how good/bad they are? Thanks! Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing timer
On 7/26/06, Zenone [EMAIL PROTECTED] wrote: But my question was, is it possible to free the channel if it rings toolong?MichelUsing this thread, is there a way to make differents rings? When receiving a call from a internal user () rings different when a external agent calls (). -- Ralph LiebessohnICQ: 74835911Skype: liebessohn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] long distance ethernet Asterisk
Two questions: We need to run Ethernet out to a really long distance 20,000ft. We have the ability to put a powered repeater in at about 12,000. We can run it using up to 4 pairs. Any recommendations on products that will reach that far? Were looking for 5 10Mbps. The products were likely looking at might be something like g.SHDSL, although Im fine with a completely proprietary solution. Any idea if it would add too much latency to run a SIP phone? TIA --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] __ ConfidentialityWarning:Thismessageandanyattachmentsareintendedonlyfortheuseoftheintendedrecipient(s), areconfidential,andmaybeprivileged.Ifyouarenottheintendedrecipient,youareherebynotifiedthatanyreview, retransmission,conversiontohardcopy,copying,circulationorotheruseofthismessageandanyattachmentsisstrictly prohibited.Ifyouarenottheintendedrecipient,pleasenotifythesenderimmediatelybyreturne-mail,anddeletethis messageandanyattachmentsfromyoursystem.Thankyou. __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager interface
There are many ways to do the screen pop, I'd like to do this way: 1. Build the manager interface as an event server, which collect agent connet events. 2. Build a Java applet with the constant connection to the event server, each agent starts the Java applet at first task of each day 3. The event server sends the connect info to the computer which the agent registed, 4. The applet launch (pop up) the web based CRM application on agent computer with the caller's information 5. Agent terminates the CRM application when the call is termianted. Tielin [EMAIL PROTECTED] 07/27/06 2:16 PM On 27 Jul 2006, at 11:47, Lee Archer wrote: This has probably been discussed before but I need to do a screen pop and I'm looking for ways to do it. I am assuming I need to use the manager interface, which is ok cos I'm using that for calling out but I'm not quite what to pick up on. There a number of ways to do this: 1) run an application on each workstation which speaks the manager protocol and pops a screen as needed. This doesn't scale easily to large numbers, you need to install an application on each workstation and need some sort of manager proxy as asterisk does not like many manager connections. 2) run an IM client on each workstation and have a central server that talks the manager protocol to asterisk, sending messages to IM clients when new calls come in. 3) have each user point their webbrowser at a web server which talks the manager protocol to asterisk and have the webpage poll the server (using AJAX) 4) embed a softphone in your application (or web page) and send calls to it. Configure the softphone to pop the screen when a call comes in. We do 4) . which you chose depends on your needs/skills. Tim. Regards Lee ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] long distance ethernet Asterisk
Check out ethernet extenders http://www.rad-direct.com/App-Ethernet-extender-copper.htm On Thu, 2006-07-27 at 15:39 -0600, Brian Vincent (C) wrote: Two questions: 1. We need to run Ethernet out to a really long distance – 20,000ft. We have the ability to put a powered repeater in at about 12,000’. We can run it using up to 4 pairs. Any recommendations on products that will reach that far? We’re looking for 5 – 10Mbps. 2. The products we’re likely looking at might be something like g.SHDSL, although I’m fine with a completely proprietary solution. Any idea if it would add too much latency to run a SIP phone? TIA --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] __ Confidentiality Warning: This message and any attachments are intended only for the use of the intended recipient(s), are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review, retransmission, conversion to hard copy, copying, circulation or other use of this message and any attachments is strictly prohibited. If you are not the intended recipient, please notify the sender immediately by return e-mail, and delete this message and any attachments from your system. Thank you. __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] long distance ethernet Asterisk
Fiber? Otherwise maybe look at cisco LRE (Long reach ethernet), but I think the limit for LRE is 5000ft (beats the heck out of regular ethernets 300ft). Last I looked LRE was very expensive. On 7/27/06, Brian Vincent (C) [EMAIL PROTECTED] wrote: Two questions: We need to run Ethernet out to a really long distance – 20,000ft. We have the ability to put a powered repeater in at about 12,000'. We can run it using up to 4 pairs. Any recommendations on products that will reach that far? We're looking for 5 – 10Mbps. The products we're likely looking at might be something like g.SHDSL, although I'm fine with a completely proprietary solution. Any idea if it would add too much latency to run a SIP phone? TIA ---Brian VincentCopper Mountain Telecom [EMAIL PROTECTED] __ConfidentialityWarning:Thismessageandanyattachmentsareintendedonlyfortheuseoftheintendedrecipient(s), areconfidential,andmaybeprivileged.Ifyouarenottheintendedrecipient,youareherebynotifiedthatanyreview,retransmission,conversiontohardcopy,copying,circulationorotheruseofthismessageandanyattachmentsisstrictly prohibited.Ifyouarenottheintendedrecipient,pleasenotifythesenderimmediatelybyreturne-mail,anddeletethismessageandanyattachmentsfromyoursystem.Thankyou.__ ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] long distance ethernet Asterisk
If you have line of sight between the points, maybe you could setup a wireless link point to point, I know some people who have done it over 3 to 5 miles range, they get 10 Mbps, (but don´t know if you could get more). just a thought Joe Pukepail escribió: Fiber? Otherwise maybe look at cisco LRE (Long reach ethernet), but I think the limit for LRE is 5000ft (beats the heck out of regular ethernets 300ft). Last I looked LRE was very expensive. On 7/27/06, Brian Vincent (C) [EMAIL PROTECTED] wrote: Two questions: We need to run Ethernet out to a really long distance – 20,000ft. We have the ability to put a powered repeater in at about 12,000'. We can run it using up to 4 pairs. Any recommendations on products that will reach that far? We're looking for 5 – 10Mbps. The products we're likely looking at might be something like g.SHDSL, although I'm fine with a completely proprietary solution. Any idea if it would add too much latency to run a SIP phone? TIA --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] __ Confidentiality Warning: This message and any attachments are intended only for the use of the intended recipient(s), are confidential, and may be privileged. If you are not the intended recipient, you are hereby notified that any review, retransmission, conversion to hard copy, copying, circulation or other use of this message and any attachments is strictly prohibited. If you are not the intended recipient, please notify the sender immediately by return e-mail, and delete this message and any attachments from your system. Thank you. __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] long distance ethernet Asterisk
I would really look towards fiber, the bandwidth and distance can easily be handled.On 7/27/06, Manrique Feoli [EMAIL PROTECTED] wrote: If you have line of sight between the points, maybe you could setup a wireless link point to point, I know some people who have done it over 3 to 5 miles range, they get 10 Mbps, (but don´t know if you could get more). just a thought Joe Pukepail escribió: Fiber? Otherwise maybe look at cisco LRE (Long reach ethernet), but I think the limit for LRE is 5000ft (beats the heck out of regular ethernets 300ft). Last I looked LRE was very expensive. On 7/27/06, Brian Vincent (C) [EMAIL PROTECTED] wrote: Two questions: We need to run Ethernet out to a really long distance – 20,000ft. We have the ability to put a powered repeater in at about 12,000'. We can run it using up to 4 pairs. Any recommendations on products that will reach that far? We're looking for 5 – 10Mbps. The products we're likely looking at might be something like g.SHDSL, although I'm fine with a completely proprietary solution. Any idea if it would add too much latency to run a SIP phone? TIA --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] __ ConfidentialityWarning:Thismessageandanyattachmentsareintendedonlyfortheuseoftheintendedrecipient(s), areconfidential,andmaybeprivileged.Ifyouarenottheintendedrecipient,youareherebynotifiedthatanyreview, retransmission,conversiontohardcopy,copying,circulationorotheruseofthismessageandanyattachmentsisstrictly prohibited.Ifyouarenottheintendedrecipient,pleasenotifythesenderimmediatelybyreturne-mail,anddeletethis messageandanyattachmentsfromyoursystem.Thankyou. __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] long distance ethernet Asterisk
Plus with fiber there's no lighting surge risk that'll burn out your equipment at both ends if the lightning hits the ground anywhere nearby.-brandonOn 7/27/06, Bruce Reeves [EMAIL PROTECTED] wrote: I would really look towards fiber, the bandwidth and distance can easily be handled.On 7/27/06, Manrique Feoli [EMAIL PROTECTED] wrote: If you have line of sight between the points, maybe you could setup a wireless link point to point, I know some people who have done it over 3 to 5 miles range, they get 10 Mbps, (but don´t know if you could get more). just a thought Joe Pukepail escribió: Fiber? Otherwise maybe look at cisco LRE (Long reach ethernet), but I think the limit for LRE is 5000ft (beats the heck out of regular ethernets 300ft). Last I looked LRE was very expensive. On 7/27/06, Brian Vincent (C) [EMAIL PROTECTED] wrote: Two questions: We need to run Ethernet out to a really long distance – 20,000ft. We have the ability to put a powered repeater in at about 12,000'. We can run it using up to 4 pairs. Any recommendations on products that will reach that far? We're looking for 5 – 10Mbps. The products we're likely looking at might be something like g.SHDSL, although I'm fine with a completely proprietary solution. Any idea if it would add too much latency to run a SIP phone? TIA --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] __ ConfidentialityWarning:Thismessageandanyattachmentsareintendedonlyfortheuseoftheintendedrecipient(s), areconfidential,andmaybeprivileged.Ifyouarenottheintendedrecipient,youareherebynotifiedthatanyreview, retransmission,conversiontohardcopy,copying,circulationorotheruseofthismessageandanyattachmentsisstrictly prohibited.Ifyouarenottheintendedrecipient,pleasenotifythesenderimmediatelybyreturne-mail,anddeletethis messageandanyattachmentsfromyoursystem.Thankyou. __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Brandon GalbraithEmail: [EMAIL PROTECTED] AIM: brandong00Voice: 630.400.6992A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Ringing timer
Use a variable that is set when the call comes in such as: Exten = s,n,Set(OUTSIDECALL=1) Then in your dial macro test for variable existence and change ring via alert info or other distinctive ring methods. It is unfortunate that it is heavily dependant on technology of the channel used. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ralph Liebessohn Sent: Thursday, July 27, 2006 5:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Ringing timer On 7/26/06, Zenone [EMAIL PROTECTED] wrote: But my question was, is it possible to free the channel if it rings too long? Michel Using this thread, is there a way to make differents rings? When receiving a call from a internal user () rings different when a external agent calls (). -- Ralph Liebessohn ICQ: 74835911 Skype: liebessohn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Getting no Audio with G729
Make sure the binary you downloaded MATCHES your machine. snip ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Getting no Audio with G729
- Original Message - From: Wasif [mailto:[EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thu, 27 Jul 2006 18:06:39 -0300 Subject: [asterisk-users] Getting no Audio with G729 Hello, Bonjour. Recently I purchased g729 codec and installed in Tribox 1.1(upgraded 1.1.1)/ Asterisk. I have pointed a DID from my carrier via SIP through g729 to asterisk. Problem is I am not getting any audio even though I am getting DTMF in asterisk. I am trying to run A2billing with asterisks. Are you behind NAT? what exactly is the callflow like for this? does the call come in and then you dial another phone? are you getting any strange messages on the Asterisk console? We're going to need some more information to track this down. Console output and a sip debug would be good. Configuration of carrier is asterisk is: [abc] allow=g729 context=c-DID dtmfmode=auto host=xxx.xxx.xxx.xxx insecure=very sendrpid=yes type=friend echo=no Any suggestions ? One final note, 'echo' is not a valid option. Thanks Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] adding a voice conversation recording on a existing PBX system
Hello all I have been recently asked a question that I dont know how to possible answer correctly. My friend who is a tech in a small company who has a Panasonic D500 PBX super hybrid system ( Out of dated) and one day his boss wants him to record all conversation between the 24 lines inbound and outbound. First of my thought / idea to him would be to see if the Panasonic system has any module to add on to allow that function. But I could not find any. So I thought of adding an asterisk in it. But that would mean nightmare since I would be needing to do something like this .. Now at the moment T1 24 lines -- PBX --- Telephone Solution T1-24 lines -- PBX -- 24 bloody x100p or similar in an asterisk --- Telephone Even though I dont think it would work since the amount of modification would be far too much. Therefore I am now asking for help and idea therefore I could give my friend a bit of help Sam ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] playing a sound into a meetme conf
Hi All, I have a problem and I'm not sure if a solution is possible without using the asterisk testing code. I am developing a volunteer translation service that users can dial into. I have a list of volunteer translators cell phone numbers stored in a mysql database along with times that they have volunteered to act as translators. That I pull from using some perl AGI scripts. A user calls, I ask which language they need help/translation with, then I put the users into a meetme conference while I call translators and play them a message asking if they're available at this time. They can refuse or accept the call. Once I get a translator that has accepted the call I connect the translator as an administrator to the meetme conference that is holding the user that is listening to music on hold. That is all working quite well with the Dialplan and AGI scripts I have set up. Problems happen when the translator drops the call midway through the conversation. i.e. Losing cell phone service. When that happens I need a way to play a message to the user to let them know that the translator has been lost and we're looking for a new one. I then need to put back the music on hold, then run deadagi scripts to find a new translator to connect to the meetme conference to help out the user. What is currently happening is that the user is left in the conference alone forever listening to MOH. I think there are two ways to do this, but I can't find out how to do either from any documentation I've found. 1. Break the user out of the meetme conf and back into the dialplan. - If I kick them from the conference they are immediately hung up on and I don't know how to stop this from happening. - There is function that is available in Asterisk 1.4 called ManagerRedirect that seems like it could do this for me, but i'd rather not try to integrate this into 1.2.10 because I fear breaking too many other things and running 1.4 (testing) just isn't an option at this time. (details here: http://bugs.digium.com/view.php?id=6508) 2. Play a message into the conference - Can I join a new pseudo channel that I've created to a meetme conf that plays a message? Does anyone know how to do this? - Can I override the MOH and stream a recorded message into the conference with only the single user in the meetme conf? Any help/ideas are appreciated. Cheers, - Simon Simon Austin http://simon.openflows.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] long distance ethernet Asterisk
I know.. I know fiber would be ideal. We have single-mode all over the place. We even have some dark, unterminated strands within 2000ft of this location it makes me want to cry. Unfortunately lighting it up isnt an option we wouldnt gain anything because we couldnt connect to anything else to get us the last stretch. Trenching 2000ft isnt an option this is National Forest land and were not allowed to do that. As far as wireless no line of sight. This location sits in a little bowl at 11,200. So what Im left with is a 400pr, 22awg out to 3000. Then we jump on 200pr, 24awg aerial cable strung on the 3rd longest high-speed quad chairlift (10,800 run). The last leg involves a short underground to another high-speed quad and down 6000. We can stick a powered repeater in the motor room of the first lift (so I guess a bit further than the original 12,000 I was thinking.) Yes, we do strange things. If youre really curious, heres a map of the campus environment we maintain: http://www.skireport.com/colorado/copper/trailmap/ --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves Sent: Thursday, July 27, 2006 4:03 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] long distance ethernet Asterisk I would really look towards fiber, the bandwidth and distance can easily be handled. On 7/27/06, Manrique Feoli [EMAIL PROTECTED] wrote: If you have line of sight between the points, maybe you could setup a wireless link point to point, I know some people who have done it over 3 to 5 miles range, they get 10 Mbps, (but don´t know if you could get more). just a thought Joe Pukepail escribió: Fiber? Otherwise maybe look at cisco LRE (Long reach ethernet), but I think the limit for LRE is 5000ft (beats the heck out of regular ethernets 300ft). Last I looked LRE was very expensive. On 7/27/06, Brian Vincent (C) [EMAIL PROTECTED] wrote: Two questions: 1. We need to run Ethernet out to a really long distance 20,000ft. We have the ability to put a powered repeater in at about 12,000'. We can run it using up to 4 pairs. Any recommendations on products that will reach that far? We're looking for 5 10Mbps. 2. The products we're likely looking at might be something like g.SHDSL, although I'm fine with a completely proprietary solution. Any idea if it would add too much latency to run a SIP phone? TIA --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] __ ConfidentialityWarning:Thismessageandanyattachmentsareintendedonlyfortheuseoftheintendedrecipient(s), areconfidential,andmaybeprivileged.Ifyouarenottheintendedrecipient,youareherebynotifiedthatanyreview, retransmission,conversiontohardcopy,copying,circulationorotheruseofthismessageandanyattachmentsisstrictly prohibited.Ifyouarenottheintendedrecipient,pleasenotifythesenderimmediatelybyreturne-mail,anddeletethis messageandanyattachmentsfromyoursystem.Thankyou. __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks __ ConfidentialityWarning:Thismessageandanyattachmentsareintendedonlyfortheuseoftheintendedrecipient(s), areconfidential,andmaybeprivileged.Ifyouarenottheintendedrecipient,youareherebynotifiedthatanyreview, retransmission,conversiontohardcopy,copying,circulationorotheruseofthismessageandanyattachmentsisstrictly prohibited.Ifyouarenottheintendedrecipient,pleasenotifythesenderimmediatelybyreturne-mail,anddeletethis messageandanyattachmentsfromyoursystem.Thankyou.
RE: [asterisk-users] adding a voice conversation recording on a existingPBX system
Id think youd want Asterisk to sit between the T1 and the D500. Pump the T1 directly in and then pump it directly out. Logic in the middle to record the call is left as an exercise for the reader. They make off-the-shelf products to do this. I dont know any names, but big call centers Ive been to in Vegas do it. --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sam Tam Sent: Thursday, July 27, 2006 4:22 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] adding a voice conversation recording on a existingPBX system Hello all I have been recently asked a question that I dont know how to possible answer correctly. My friend who is a tech in a small company who has a Panasonic D500 PBX super hybrid system ( Out of dated) and one day his boss wants him to record all conversation between the 24 lines inbound and outbound. First of my thought / idea to him would be to see if the Panasonic system has any module to add on to allow that function. But I could not find any. So I thought of adding an asterisk in it. But that would mean nightmare since I would be needing to do something like this .. Now at the moment T1 24 lines -- PBX --- Telephone Solution T1-24 lines -- PBX -- 24 bloody x100p or similar in an asterisk --- Telephone Even though I dont think it would work since the amount of modification would be far too much. Therefore I am now asking for help and idea therefore I could give my friend a bit of help Sam __ ConfidentialityWarning:Thismessageandanyattachmentsareintendedonlyfortheuseoftheintendedrecipient(s), areconfidential,andmaybeprivileged.Ifyouarenottheintendedrecipient,youareherebynotifiedthatanyreview, retransmission,conversiontohardcopy,copying,circulationorotheruseofthismessageandanyattachmentsisstrictly prohibited.Ifyouarenottheintendedrecipient,pleasenotifythesenderimmediatelybyreturne-mail,anddeletethis messageandanyattachmentsfromyoursystem.Thankyou. __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SNOM 360
Hi List,Does anyone know how to set up QoS on the SNOM 360 ? Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Getting no Audio with G729
Hi again, Asterisk was not behind the NAT and I downloaded correct platform of codec. I solved my problem by changing the prompts into G729 format. And it works fine now. Now I need to know about a utility which can convert all ulaw audio prompts into g729 prompts in bulk. Or is there any was Asterisk can convert ulaw prompts to G729 prompts by itself during call. Thanks , -Original Message- From: Wasif [mailto:[EMAIL PROTECTED] Sent: Thursday, July 27, 2006 5:07 PM To: 'asterisk-users@lists.digium.com' Subject: Getting no Audio with G729 Hello, Recently I purchased g729 codec and installed in Tribox 1.1(upgraded 1.1.1)/ Asterisk. I have pointed a DID from my carrier via SIP through g729 to asterisk. Problem is I am not getting any audio even though I am getting DTMF in asterisk. I am trying to run A2billing with asterisks. Configuration of carrier is asterisk is: [abc] allow=g729 context=c-DID dtmfmode=auto host=xxx.xxx.xxx.xxx insecure=very sendrpid=yes type=friend echo=no Any suggestions ? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 Schedule and Features/Changes
Hi all, Asterisk 1.4 was originally scheduled to be released early July 2006. Is there an update on the expected release of this version? Also is there a changelog or feature list available that lists the differences over 1.2? TIA, Max -- Max Clark http://www.clarksys.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gxp-2000 configure line appearances
Hello Cavanna,, * Cavanna, Richard [EMAIL PROTECTED] [27-07-06 15:59]: The real thing that would help is a complete list of the configurable comands on the latest firmware so I can create the config file. try that config file, works perfectly for me. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ## Configuration template for GXP-2000 firmware version 1.0.2.13 ## ## Advanced/System-wide Options ## # Admin password for web interface P2 = admin # Silence Suppression. 0 - no, 1 - yes P50 = 1 # Voice Frames per TX (up to 10/20/32/64 frames for G711/G726/G723/other codecs respectively) P37 = 2 # Layer 3 QoS (IP Diff-Serv or Precedence value for RTP) P38 = 48 # Layer 2 QoS. 802.1Q/VLAN Tag (VLAN classification for RTP) P51 = 0 # Layer 2 QoS. 802.1p priority value (0 - 7) P87 = 0 # No Key Entry Timeout. Default - 4 seconds. P85 = 4 # Use # as Dial Key (if set to Yes, # will function as the (Re-)Dial key). 0 - no, 1 - yes P72 = 1 # Local RTP port (1024-65535, default 5004) P39 = 5004 # Use Random Port. 0 - no, 1 - yes P78 = 0 # Keep-alive interval (in seconds. default 20 seconds) P84 = 20 # Use NAT IP. This will enable our SIP client to use this IP in the SIP message. Example 64.3.153.50. P101 = # STUN server P76 = #- # Firmware Upgrade #- # Firmware Upgrade. 0 - TFTP Upgrade, 1 - HTTP Upgrade. P212 = 0 # Firmware Server Path P192 = 192.168.0.251 # Config Server Path P237 = 192.168.0.251 # Firmware File Prefix P232 = # Firmware File Postfix P233 = # Config File Prefix P234 = # Config File Postfix P235 = # Allow DHCP Option 66 to override server. 0 - No, 1 - Yes. Default is No. # When set to Yes(1), it will override the configured provision path and method. P145 = 0 # Automatic Upgrade. 0 - No, 1 - Yes (checking every defined days). Default is No. P194 = 1 # Check for new firmware every () minutes, unit is in minute, default is 7 days. P193 = 10080 # Use firmware pre/postfix to determine if f/w is required # 0 = Always Check for New Firmware # 1 = Check New Firmware only when F/W pre/suffix changes P238 = 0 # DTMF Payload Type P79 = 101 # Syslog Server (name of the server, max length is 64 charactors) P207 = 192.168.0.251 # Syslog Level (Default setting is NONE) # 0 - NONE, 1 - DEBUG, 2 - INFO, 3 - WARNING, 4 - ERROR P208 = 0 # NTP Server P30 = 192.168.0.251 # Allow DHCP Option 42 to override NTP server. 0 - No, 1 - Yes. Default is No. # When set to Yes(1), it will override the configured NTP server. P144 = 0 # Distinctive Ring Tone # Use custom ring tone 1 if incoming caller ID is the following: P105 = # Use custom ring tone 2 if incoming caller ID is the following: P106 = # Use custom ring tone 3 if incoming caller ID is the following: P107 = # Disable Call Waiting. 0 - no, 1 - yes P91 = 0 # Lock Keypad Update. 0 - no, 1 - yes P88 = 0 # Primary Account (Account 1) Settings # Account Active (In Use). 0 - no, 1 - yes P271 = 1 # Account Name P270 = # SIP Server P47 = sip.mycompany.com # Outbound Proxy P48 = proxy.mycompany.com # SIP User ID P35 = 8000 # Authenticate ID P36 = 8000 # Authenticate password P34 = # Display Name (John Doe) P3 = # Use DNS SRV. 0 - No, 1 - Yes. P103 = 0 # SIP User ID is phone number. 0 - no, 1 - yes P63 = 0 # SIP Registration. 0 - no, 1 - yes P31 = 1 # Unregister On Reboot. 0 - no, 1 - yes P81 = 0 # Register Expiration (in minutes. default 1 hour, max 45 days) P32 = 60 # Local SIP port (default 5060) P40 = 5060 # SIP T1 Timeout. RFC 3261 T1 value (RTT estimate) # 50 - 0.5 sec, 100 - 1 sec, 200 - 2 sec. Default 100. P209 = 100 # SIP T2 Interval. RFC 3261 T2 value. The maximum retransmit interval for non-INVITE requests and INVITE responses. # 200 - 2 sec, 400 - 4 sec, 800 - 8 sec. Default 400. P250 = 400 # NAT Traversal. 0 - yes, 1 - no, 2 - No, but send keep-alive P52 = 0 # SUBSCRIBE for MWI. (Whether or not send SUBSCRIBE for Message Waiting Indication) 0 - No, 1 - Yes. P99 = 1 # Proxy-Require (A SIP extension to enable firewall penetration) P197 = # Voice Mail UserID (User ID/extension for 3rd party voice mail system) P33 = 88 # Send DTMF. 0 - in audio, 1 - via RTP, 2 - via SIP INFO P73 = 2 # Early Dial (use Yes
